Vicidial + Asterisk 1.4.27.1-vici

trandos

Neuer User
Mitglied seit
21 Sep 2010
Beiträge
8
Punkte für Reaktionen
0
Punkte
0
hi alle zusammen.

ich habe ein Problem. Ich habe einen Vicidialserver aufgesetzt das ist eine Callcenter Open source software die mit asterisk arbeitet.

Ich lebe in griechenland und habe einen griechischen Sip provider.


ich habe ueber vicidial meine sip configuration eingegeben welche automatisch in asterisk weitergegeben wird siehe:

registrierungsstring ist ok denn ich erhalte in sip sho registry das er registriert ist.

Account Entry:
Code:
[viva]
type=peer
username=302117706001
secret=XXXXXX
fromuser=302117706001
host=viva.gr
dtmfmode=rfc2833
fromdomain=viva.gr
context=trunkinbound
nat=yes
insecure=very
dtmfmode=rfc2833
disallow=all
disallow=g729
canreinvite=no
allow=all
qualify=1000 

global string:VIVA=SIP/viva

dialplan: 
exten => _9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXX,2,Dial(${VIVA}/${EXTEN:1},,tTo)
exten => _9XXXXXXXXXX,3,Hangup
so weit so gut.

ich benutze xlite als siphone wo ich mein phone 103 habe. dieser verbindet auch mit vicidial.

sobald ich auf waehlen druecke erhalte ich folgendes in cli.
Code:
: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 21 Sep 2010 20:43:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


---
[Sep 21 16:43:13]
<--- SIP read from 192.168.1.59:17892 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK644ad82a;rport=5060
Contact: <sip:192.168.1.59:17892>
To: <sip:[email protected]:17892;rinstance=ac2a95433cd88040;cpd=on>;tag=1858b20
From: "asterisk"<sip:[email protected]>;tag=as2ab4829c
Call-ID: [email protected]
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INO
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0


<------------->
[Sep 21 16:43:13] --- (12 headers 0 lines) ---
[Sep 21 16:43:13] Really destroying SIP dialog '[email protected]' Method: OPTIONS
[Sep 21 16:43:18]
<--- SIP read from 192.168.1.59:17892 --->



<------------->
[Sep 21 16:43:19]
<--- SIP read from 83.235.24.86:5060 --->

<------------->
[Sep 21 16:43:24]   == Parsing '/etc/asterisk/manager.conf': [Sep 21 16:43:24] ound
[Sep 21 16:43:24]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 21 16:43:24]     -- Executing [8600051@default:1] MeetMe("Local/8600051@deault-d14b,2", "8600051|F") in new stack
[Sep 21 16:43:24]        > Channel Local/8600051@default-d14b,1 was answered.
[Sep 21 16:43:24]   == Starting Local/8600051@default-d14b,1 at default,91695971530,1 failed so falling back to exten 's'
[Sep 21 16:43:24]   == Starting Local/8600051@default-d14b,1 at default,s,1 stil failed so falling back to context 'default'
[Sep 21 16:43:24]     -- Sent into invalid extension 's' in context 'default' o Local/8600051@default-d14b,1
[Sep 21 16:43:24]     -- Executing [i@default:1] Playback("Local/8600051@defaul-d14b,1", "invalid") in new stack
[Sep 21 16:43:24]     -- <Local/8600051@default-d14b,1> Playing 'invalid' (langage 'en')
[Sep 21 16:43:24] WARNING[11136]: file.c:1292 waitstream_core: Unexpected contrl subclass '-1'
[Sep 21 16:43:24] WARNING[11136]: file.c:1292 waitstream_core: Unexpected contrl subclass '-1'
[Sep 21 16:43:28]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 21 16:43:28]   == Auto fallthrough, channel 'Local/8600051@default-d14b,1' status is 'UNKNOWN'
[Sep 21 16:43:28]     -- Executing [h@default:1] DeadAGI("Local/8600051@default-d14b,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Sep 21 16:43:28]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Sep 21 16:43:28]   == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-d14b,2'
[Sep 21 16:43:28]     -- Executing [h@default:1] DeadAGI("Local/8600051@default-d14b,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Sep 21 16:43:28]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0

wenn ich manuell wahle erscheint folgende meldung.
Code:
"agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----17-----BUSY----------") in new stack
[Sep 21 16:51:04]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----17-----BUSY---------- completed, returning 0
[Sep 21 16:51:04] Scheduling destruction of SIP dialog 'YmEzMTk4MWZjOThhMGQ3MjA0MjgxZTY4ZDA5NGEwYzQ.' in 32000 ms (Method: INVITE)
[Sep 21 16:51:04]
<--- Reliably Transmitting (NAT) to 192.168.1.59:17892 --->
SIP/2.0 486 Busy here
Via: SIP/2.0/UDP 192.168.1.59:17892;branch=z9hG4bK-d8754z-bf3701135c5e8759-1---d8754z-;received=192.168.1.59;rport=17892
From: "cc103"<sip:[email protected]>;tag=264a4351
To: "92114046765"<sip:[email protected]>;tag=as21a4e4fc
Call-ID: YmEzMTk4MWZjOThhMGQ3MjA0MjgxZTY4ZDA5NGEwYzQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
[Sep 21 16:51:04]
<--- SIP read from 192.168.1.59:17892 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.59:17892;branch=z9hG4bK-d8754z-bf3701135c5e8759-1---d8754z-;rport
To: "92114046765"<sip:[email protected]>;tag=as21a4e4fc
From: "cc103"<sip:[email protected]>;tag=264a4351
Call-ID: YmEzMTk4MWZjOThhMGQ3MjA0MjgxZTY4ZDA5NGEwYzQ.
CSeq: 2 ACK
Content-Length: 0


<------------->
[Sep 21 16:51:04] --- (7 headers 0 lines) ---
[Sep 21 16:51:04] Really destroying SIP dialog '[email protected]' Method: INVITE
go*CLI>

vicidial forum konnte mir leider nicht helfen ! ich denke es liegt an der extensio aber ich bin natuerlich kein profi ! koennt ihr mir bitte helfen ??
 
Hallo und willkommen im Forum,

es wird versucht, im Context default eine Nummer zu wählen, zu der es keine passende exten gibt. Ist das was Du als "dialplan" bezeichnest der Inhalt von [default] oder ein anderer Context? Wenn es ein anderer Context ist, solltest Du dem zum Wählen verwendeten Tool beibringen, diesen zu nutzen.

Außerdem ist Dein Dial falsch, ausgeschrieben würde das bei Dir Dial(SIP/viva/${EXTEN:1}) ergeben.
Die Syntax für abgehende Anrufe über SIP ist Dial(SIP/${EXTEN:1}@viva)
Das lässt sich mit der globalen Variable so nicht darstellen.
Die Option t solltest Du bei abgehenden Gesprächen raus nehmen.
 
hi rentier danke erstamal fuer deine antwort.

so wie ich die einstellungen in asterisk sehe gibt es eine extensions.conf die auf extensions.conf_vicidial verweist. d.h was ich in vicidial eingebe wird automatisch in die extensions.conf uebertragen. das gleiche gilt fuer die sip.conf. welche auf sip.conf_vicidial verweist siehe:
Code:
[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp                             ; Console interface for demo
;TRUNK=Zap/r1                                    ; Trunk interface
;TRUNKX=Zap/r2					; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:[email protected]:4569	; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1:[email protected]:4569	; IAX trunk interface
;TRUNKBINFONE=IAX2/1112223333:[email protected]	; IAX trunk interface
;SIPtrunk=SIP/1234:[email protected]	; SIP trunk
[COLOR="Lime"]

#include extensions-vicidial.conf

[trunkinbound]
; DID call routing process
exten => _X.,1,AGI(agi-DID_route.agi)

; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})


[loopback-no-log]
; This context is to accept calls that have already been logged in another context in Vicidial 
; and has been sent through one of the loopbacks. This is why this context is missing the h extension.
; Do not put any extensions in this context unless you specifically understand what this means.

;exten => _91NXXNXXXXXX,1,Dial(${TRUNKX}/${EXTEN:1},,To)
;exten => _91NXXNXXXXXX,n,Hangup
; special Canadian PRI callerIDname settings FOR USE IN LOOPBACK CONTEXT ONLY
;exten => _91NXXNXXXXXX,1,Set(CALLERID(name)="ACME Widgets")
;exten => _91NXXNXXXXXX,n,AGI(agi-CANADA_PRI_CIDname.agi)
;exten => _91NXXNXXXXXX,n,Dial(${TRUNKX}/${EXTEN:1},,To)
;exten => _91NXXNXXXXXX,n,Hangup

exten => _999XX11112,1,Wait(2)
exten => _999XX11112,n,Answer
exten => _999XX11112,n,Playback(ss-noservice)
exten => _999XX11112,n,Playback(vm-goodbye)
exten => _999XX11112,n,Hangup



[default]
include => vicidial-auto

; VICI-GROUP DIRECT SUPPORT LINE (VICIHELP[84244357])
exten => _84244XXX,1,Dial(IAX2/vicihelp/${EXTEN:5})

; Local agent alert extensions
exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
; Local blind monitoring
exten => _08600XXX,1,Dial(${TRUNKblind}/6${EXTEN:1},55,To)


; playback of recorded prompts
exten => _851XXXXX,1,Answer
exten => _851XXXXX,2,Playback(${EXTEN})
exten => _851XXXXX,3,Hangup

; this is used for playing a message to an answering machine forwarded from AMD in VICIDIAL
exten => _7851XXXXX,1,WaitForSilence(2000,2) ; AMD got machine.  leave message after recording
exten => _7851XXXXX,2,Playback(${EXTEN:1})
exten => _7851XXXXX,3,AGI(VD_amd_post.agi,${EXTEN:1})
exten => _7851XXXXX,4,Hangup


; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})


; Example phone extensions

; Extension 2000 Sipura/Linksys ATA line 1
;exten => 2000,1,Dial(sip/spa2000,30,to)   ; Ring, 30 secs max
;exten => 2000,2,Voicemail,u2000           ; Send to voicemail...
; Extension 2001 Sipura/Linksys ATA line 2
;exten => 2001,1,Dial(sip/spa2001,30,to)   ; Ring, 30 secs max
;exten => 2001,2,Voicemail,u2001           ; Send to voicemail...
; Extension 2102 rings Grandstream phone
;exten => 2102,1,Dial(sip/gs102,30,to)    ; Ring, 30 secs max
;exten => 2102,2,Voicemail,u2102          ; Send to voicemail...
; Extension 401 rings the firefly softphone
;exten => 401,1,Dial((IAX2/firefly01@firefly01/s||t)
;exten => 401,2,Hangup

; 100-350 phone extensions now auto-generated
; extensions for other SIP and IAX call center phones
;   cc100-cc150 SIP Phones
;exten => _1[0-5]X,1,Dial(sip/cc${EXTEN},20,to)
;   cc300-cc350 IAX Phones
;exten => _3[0-5]X,1,Dial(IAX2/cc${EXTEN},20,to)

; extensions if using a T1 channelbank
;exten => _19XX,1,Dial(Zap/${EXTEN:2},30,o)
;exten => _19XX,2,Hangup

; Extension 4001 rings Zap phone (this example for FXS on Zap port 1)
;exten => 4001,1,Dial(Zap/1,30,o)	; ring Zap device 1
;exten => 4001,2,Voicemail,u4001         ; Send to voicemail...


; # timeout invalid rules
exten => #,1,Playback(invalid)              ; "Thanks for trying the demo"
exten => #,2,Hangup                     ; Hang them up.
exten => t,1,Goto(#,1)                  ; If they take too long, give up
exten => i,1,Playback(invalid)          ; "That's not valid, try again"


; Extensions for performance testing 
exten => _91999NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91999NXXXXXX,2,Dial(${TRUNKloop}/${EXTEN:2},,tTo)
exten => _91999NXXXXXX,3,Hangup
exten => 999999999999,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 999999999999,2,Dial(${TRUNKloop}/${EXTEN:1},,tTo)
exten => 999999999999,3,Hangup

; This is a loopback dialaround to allow for hearing of ringing for 3way calls
exten => _881NXXNXXXXXX,1,Answer
exten => _881NXXNXXXXXX,2,Dial(${TRUNKloop}/9${EXTEN:2},,To)
exten => _881NXXNXXXXXX,3,Hangup

; Vtiger fax and email log extensions
exten => _9118XXXXXXXX,1,Dial(${TRUNKblind}/9998818112,55,to)
exten => _9119XXXXXXXX,1,Dial(${TRUNKblind}/9998819112,55,to)


; CARRIER DIALING EXTENSIONS, USE THE ADMIN INTERFACE TO PROGRAM THESE
; dial an 800 outbound number
;exten => _91800NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
;exten => _91800NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,To)
;exten => _91800NXXXXXX,3,Hangup
;exten => _91888NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
;exten => _91888NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,To)
;exten => _91888NXXXXXX,3,Hangup
;exten => _91877NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
;exten => _91877NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,To)
;exten => _91877NXXXXXX,3,Hangup
;exten => _91866NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
;exten => _91866NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,To)
;exten => _91866NXXXXXX,3,Hangup

; dial a long distance outbound number
; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls
;exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
;exten => _91NXXNXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,To)
;exten => _91NXXNXXXXXX,3,Hangup

; dial a local outbound number (modified because of only LD T1)
;exten => _9NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
;exten => _9NXXXXXX,2,Dial(${TRUNK}/1727${EXTEN:1},,To)
;exten => _9NXXXXXX,3,Hangup

; dial a local 727 outbound number with area code
;exten => _9727NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
;exten => _9727NXXXXXX,2,Dial(${TRUNK}/1${EXTEN:1},,To)
;exten => _9727NXXXXXX,3,Hangup

; dial a long distance outbound number to the UK
; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls, 
;exten => _901144XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
;exten => _901144XXXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},55,To)
;exten => _901144XXXXXXXXXX,3,Hangup

; dial a long distance outbound number to Australia
; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls, 
;exten => _901161XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
;exten => _901161XXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,To)
;exten => _901161XXXXXXXXX,3,Hangup

; dial a long distance outbound number through BINFONE
; exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91NXXNXXXXXX,2,Dial(${TRUNKIAX}/${EXTEN:1},55,To)
; exten => _91NXXNXXXXXX,3,Hangup
; dial a long distance outbound number through a SIP provider
; exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91NXXNXXXXXX,2,Dial(sip/${EXTEN:1}@SIPtrunk,55,o)
; exten => _91NXXNXXXXXX,3,Hangup
; special extensions for North America to catch invalid phone numbers
; exten => _91XXX[0-1]XXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXX[0-1]XXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91XXX[0-1]XXXXXX,n,Hangup
; exten => _91[0-1]XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91[0-1]XXXXXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91[0-1]XXXXXXXXX,n,Hangup
; exten => _91XXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91XXXXXX,n,Hangup
; exten => _91XXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91XXXXXXX,n,Hangup
; exten => _91XXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91XXXXXXXX,n,Hangup
; exten => _91XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91XXXXXXXXX,n,Hangup
; exten => _91XXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91XXXXXXXXXXX,n,Hangup
; exten => _91XXXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91XXXXXXXXXXXX,n,Hangup
; exten => _91XXXXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXXXXXXX,n,Dial(${TRUNKloop}/8889990011112,,to)
; exten => _91XXXXXXXXXXXXX,n,Hangup
; dial a USA long distance outbound number through the loopback-no-log context
; exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91NXXNXXXXXX,2,Dial(${TRUNKloop}/888${EXTEN:2},55,o)
; exten => _91NXXNXXXXXX,3,Hangup
;exten => 888NXXNXXXXXX,1,Goto(loopback-no-log,91${EXTEN:3},1)

exten => 8889990011112,1,Goto(loopback-no-log,9990011112,1)


; Inbound call from BINFONE
; exten => 1112223333,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => 1112223333,2,Dial(sip/gs102,55,o)
; exten => 1112223333,3,Hangup

; Extension 7275551212 - Inbound local number from PRI with 10 digit delivery
;exten => 7275551212,1,Ringing
;exten => 7275551212,2,Wait(1)
;exten => 7275551212,3,AGI(agi://127.0.0.1:4577/call_log--fullCID--${EXTEN}-----${CALLERID(all)}-----${CALLERID(num)}-----${CALLERID(name)})
;exten => 7275551212,4,Answer
;exten => 7275551212,5,Dial(sip/spa2000&sip/spa2001,30,To)
;exten => 7275551212,6,Voicemail,u2000

; parameters for call_inbound.agi (7 fields separated by five dashes "-----"):
; 1. the extension of the phone to ring as defined in the asterisk.phones table
; 2. the phone number that was called, for the live_inbound/_log entry
; 3. a text description of the number that was called in
; 4-7. optional fields, they are also passed as fields in the GUI to web browser
; This is not part of VICIDIAL, it is for astGUIclient agent use only

; Extension 3429 - Inbound 800 number (1-800-555-3429) example of RBS T1
;    with 10 digit ANI and 4 digit DNIS star separated
;exten => _**3429,1,Ringing
;exten => _**3429,2,AGI(agi://127.0.0.1:4577/call_log)
;exten => _**3429,3,AGI(call_inbound.agi,spa2000-----8005553429-----Inbound 800-----x-----y-----z-----w)
;exten => _**3429,4,Answer
;exten => _**3429,5,Dial(sip/spa2000&sip/spa2001,30,to)
;exten => _**3429,6,Voicemail,u2000
; Extension 3429 - with ANI [callerID]
;exten => _*NXXNXXXXXX*3429,1,Ringing
;exten => _*NXXNXXXXXX*3429,2,AGI(agi://127.0.0.1:4577/call_log)
;exten => _*NXXNXXXXXX*3429,3,AGI(call_inbound.agi,spa2000-----8005553429-----Inbound 800-----x-----y-----z-----w)
;exten => _*NXXNXXXXXX*3429,4,Answer
;exten => _*NXXNXXXXXX*3429,5,Dial(sip/spa2000&sip/spa2001,30,to)
;exten => _*NXXNXXXXXX*3429,6,Voicemail,u2000


; parameters for agi-VDAD_ALL_inbound.agi (upto 12 fields separated by five dashes "-----"):
; Below are the parameters needed for the script to be run properly
; 1. the method of call handling for the script:
; 	- CID - 	CID received, add record with phone number
; 	- CIDLOOKUP - 	Lookup CID to find record in whole system
; 	- CIDLOOKUPRL -	Restrict lookup to one list
; 	- CIDLOOKUPRC -	Restrict lookup to one campaign's lists
;     - CLOSER -      Closer calls from VICIDIAL fronters
; 	- ANI - 	ANI received, add record with phone number (based on RBS T1s)
; 	- ANILOOKUP - 	Lookup ANI to find record in whole system
; 	- ANILOOKUPRL -	Restrict lookup to one list
; 	- ANILOOKUPRC -	Restrict lookup to one campaign's lists
; 	- VID -		Add record with Vendor Lead Code received as argument 12
; 	- VIDLOOKUP - 	Lookup Vendor Lead Code received as argument 12 to find record in whole system
; 	- VIDLOOKUPRL -	Restrict lookup to one list (argument 12)
; 	- VIDLOOKUPRC -	Restrict lookup to one campaign's lists (argument 12)
; 	- VIDPROMPT - 	Prompt Vendor Lead Code to User with IVR to add record with Vendor Lead Code
; 	- VIDPROMPTLOOKUP - 	Prompt Vendor Lead Code to User with IVR to find record in whole system
; 	- VIDPROMPTLOOKUPRL -	Restrict lookup to one list
; 	- VIDPROMPTLOOKUPRC -	Restrict lookup to one campaign's lists
; 	- 3DIGITID - 	Enter 3 digit code to go to agent
; 	- 4DIGITID - 	Enter 4 digit code to go to agent
; 	- XDIGITID - 	Enter X digit code to go to agent(variable, i.e. 9DIGITID, 12DIGITID, etc...)
; 2. the method of searching for an available agent:
; 	- LO - Load Balance Overflow only (priority to home server)
; 	- LB - <default> Load Balance total system
; 	- SO - Home server only
; 3. the full name of the IN GROUP to be used in vicidial for the inbound call
; 4. the phone number that was called, for the log entry
; 5. the callerID or lead_id of the person that called(usually overridden)
; 6. the park extension audio file name if used
; 7. the status of the call initially(usually not used)
; 8. the list_id to insert the new lead under if it is new (and CID/ANI available)
; 9. the phone dialing code to insert with the new lead if new (and CID/ANI available)
; 10. the campaign_id to search within lists if CIDLOOKUPRC
; 11. the user to queue the call to for AGENTDIRECT in-group calls
; 12. vendor_lead_code if external mechanism like custom IVR is used to prompt user for ID
;
; inbound VICIDIAL call with CID delivery through T1 PRI
;exten => 1234,1,Answer                  ; Answer the line
;exten => 1234,2,AGI(agi-VDAD_ALL_inbound.agi,CID-----LB-----CL_GALLERIA-----7274515134-----Closer-----park----------999-----1)
;exten => 1234,3,Hangup

; inbound VICIDIAL transfer calls [can arrive through PRI T1 crossover, IAX or SIP channel]
exten => _90009.,1,Answer                  ; Answer the line
exten => _90009.,2,Dial(${TRUNKloop}/9${EXTEN},,to)
exten => _90009.,3,Hangup
exten => _990009.,1,Answer                  ; Answer the line, Sometimes needs to be removed
exten => _990009.,2,AGI(agi-VDAD_ALL_inbound.agi,CLOSER-----LB-----CL_TESTCAMP-----7275551212-----Closer-----park----------999-----1)
exten => _990009.,3,Hangup
; DID forwarded calls
exten => _99909*.,1,Answer
exten => _99909*.,2,AGI(agi-VDAD_ALL_inbound.agi)
exten => _99909*.,3,Hangup


; barge monitoring extension
exten => 8159,1,ZapBarge
exten => 8159,2,Hangup

; ZapBarge direct channel extensions
exten => _86120XX,1,ZapBarge(${EXTEN:5})


exten => _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})
exten => _X48600XXX,2,Hangup

exten => _X38600XXX,1,MeetMeAdmin(${EXTEN:2},t,${EXTEN:0:1})
exten => _X38600XXX,2,Hangup

exten => _X28600XXX,1,MeetMeAdmin(${EXTEN:2},m,${EXTEN:0:1})
exten => _X28600XXX,2,Hangup

exten => _X18600XXX,1,MeetMeAdmin(${EXTEN:2},M,${EXTEN:0:1})
exten => _X18600XXX,2,Hangup

exten => _55558600XXX,1,MeetMeAdmin(${EXTEN:4},K)
exten => _55558600XXX,2,Hangup
exten => 8300,1,Hangup

; astGUIclient conferences
exten => _86000[0-4]X,1,Meetme,${EXTEN}|q
; VICIDIAL conferences
exten => _86000[5-9]X,1,Meetme,${EXTEN}|F
exten => _8600[1-2]XX,1,Meetme,${EXTEN}|F
; quiet entry and leaving conferences for VICIDIAL (inbound announce and SendDTMF)
exten => _78600XXX,1,Meetme,${EXTEN:1}|Fq
; quiet monitor-only extensions for meetme rooms (for room managers)
exten => _68600XXX,1,Meetme,${EXTEN:1}|Fmq
; quiet monitor-only entry and leaving conferences for VICIDIAL (recording)
exten => _58600XXX,1,Meetme,${EXTEN:1}|Fmq

; voicelab exten
exten => _86009XX,1,Meetme,${EXTEN}|Fmq
; voicelab exten moderator
exten => _986009XX,1,Meetme,${EXTEN:1}



; park channel for client GUI parking, hangup after 30 minutes
;    create a GSM formatted audio file named "park.gsm" that is 30 minutes long
;    and put it in /var/lib/asterisk/sounds
exten => 8301,1,Answer
exten => 8301,2,AGI(park_CID.agi)
exten => 8301,3,Playback(park)
exten => 8301,4,Hangup 
exten => 8303,1,Answer
exten => 8303,2,AGI(park_CID.agi)
exten => 8303,3,Playback(conf)
exten => 8303,4,Hangup 

; park channel for client GUI conferencing, hangup after 30 minutes
;    create a GSM formatted audio file named "conf.gsm" that is 30 minutes long
;    and put it in /var/lib/asterisk/sounds
exten => 8302,1,Answer
exten => 8302,2,Playback(conf)
exten => 8302,3,Hangup

exten => 8304,1,Answer
exten => 8304,2,Playback(ding)
exten => 8304,3,Hangup

; default audio for safe harbor 2-second-after-hello message then hangup
;    create a GSM formatted audio file complies with safe harbor rules
;    and put it in /var/lib/asterisk/sounds then change filename below
exten => 8307,1,Answer
exten => 8307,2,Playback(vm-goodbye)
exten => 8307,3,Hangup

; this is used for playing a message to an answering machine forwarded from AMD in VICIDIAL
exten => 8320,1,AGI(VD_amd.agi,${EXTEN}-----YES)
exten => 8320,2,Hangup
exten => _8320*.,1,AGI(VD_amd.agi,${EXTEN}-----YES)
exten => _8320*.,2,Hangup

; use for selective CallerID hangup by area code(hard-coded)
exten => 8352,1,AGI(agi-VDADselective_CID_hangup.agi,${EXTEN})
exten => 8352,2,Playback(safe_harbor)
exten => 8352,3,Hangup

; this is used for sending DTMF signals within conference calls, the client app
;    sends the digits to be played in the callerID field
;    sound files must be placed in /var/lib/asterisk/sounds
exten => 8500998,1,Answer
exten => 8500998,2,Playback(silence)
exten => 8500998,3,AGI(agi-dtmf.agi)
exten => 8500998,4,Hangup

; multi-remote-monitor entry extensions
exten => 8162,1,Dial(${TRUNKblind}/34567890123456789,55,to)

exten => 34567890123456789,1,Answer
exten => 34567890123456789,2,Goto(monitor,s,1)

;#### VDAD STANDARD TRANSFER ENTRIES ####
; Below are the parameters needed for the agi-VDAD_ALL_outbound.agi script to be run properly
; 1. the method of call handling for the script:
; 	- NORMAL -	 	<default> Standard outbound routing to agent
; 	- TEST - 		For performance testing only
; 	- BROADCAST -	For no-agent broadcast dialing
; 	- SURVEY -		For survery question then on to agent
; 	- REMINDER -	Reminder campaign
; 	- REMINDX -		Reminder with transfer to agent
; 2. the method of searching for an available agent:
; 	- LB - <default> Load Balance total system
; 	- LO - Load Balance Overflow only (priority to home server)
; 	- SO - Home server only
; 3. the sound file to play when doing a SURVEY, REMINDER, REMINDX campaign
; 4. the acceptible dtmf digits for a SURVEY
; 5. the out-opt digit for a SURVEY (must be in the digit map)
; 6. the sound file to play for a SURVEY when transfering to an agent
; 7. the sound file to play for a SURVEY when DNCing the call
; 8. OPTIN or OPTOUT: if OPTIN call is only sent to agent with button press
;     if OPTOUT call is sent to agent if no button press at all
; 9. the status that is use for a SURVEY when someone opts out
;     if the status is DNC it will also add them to the internal dnc table

; VICIDIAL_auto_dialer transfer script for no-agent campaigns:
exten => 8364,1,Playback(sip-silence)
exten => 8364,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8364,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8364,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8364,5,Hangup

; VICIDIAL_auto_dialer transfer script:
exten => 8365,1,Playback(sip-silence)
exten => 8365,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8365,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----SO)
exten => 8365,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----SO)
exten => 8365,5,Hangup

; VICIDIAL_auto_dialer transfer script SURVEY at beginning:
exten => 8366,1,Playback(sip-silence)
exten => 8366,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8366,3,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8366,4,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8366,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balance Overflow:
exten => 8367,1,Playback(sip-silence)
exten => 8367,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8367,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LO)
exten => 8367,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LO)
exten => 8367,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balanced:
exten => 8368,1,Playback(sip-silence)
exten => 8368,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8368,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,5,Hangup

; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten => 8369,1,Playback(sip-silence)
exten => 8369,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8369,3,AMD(2000|2000|1000|5000|120|50|4|256) 
exten => 8369,4,AGI(VD_amd.agi,${EXTEN})
exten => 8369,5,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8369,6,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8369,7,Hangup

; VICIDIAL auto-dial reminder script
exten => 8372,1,Playback(sip-silence)
exten => 8372,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8372,3,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,4,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,5,Hangup

; VICIDIAL SURVEY transfer script AMD with Load Balanced:
exten => 8373,1,Playback(sip-silence)
exten => 8373,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8373,3,AMD(2000|2000|1000|5000|120|50|4|256) 
exten => 8373,4,AGI(VD_amd.agi,${EXTEN})
exten => 8373,5,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8373,6,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8373,7,Hangup

; VICIDIAL SURVEY transfer script with Cepstral names:
exten => 8374,1,Playback(sip-silence)
exten => 8374,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8374,3,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMPCEP-----LB)
exten => 8374,4,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMPCEP-----LB)
exten => 8374,5,Hangup

; VICIDIAL SURVEY transfer script AMD with Cepstral variables:
exten => 8375,1,Playback(sip-silence)
exten => 8375,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8375,3,AMD(2000|2000|1000|5000|120|50|4|256)
exten => 8375,4,AGI(VD_amd.agi,${EXTEN})
exten => 8375,5,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMPCEP-----LB)
exten => 8375,6,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMPCEP-----LB)
exten => 8375,7,Hangup



; PERFORMANCE TESTING
exten => _999XXXXXX1,1,Answer
exten => _999XXXXXX1,2,Wait(2)
exten => _999XXXXXX1,3,Playback(vicidial-welcome)
exten => _999XXXXXX1,4,Hangup

exten => _999XX11112,1,Wait(2)
exten => _999XX11112,2,Answer
exten => _999XX11112,3,Playback(ss-noservice)
exten => _999XX11112,4,Playback(vm-goodbye)
exten => _999XX11112,5,Hangup

exten => _999XX18112,1,Wait(2)
exten => _999XX18112,2,Answer
exten => _999XX18112,3,Playback(vtiger-fax)
exten => _999XX18112,4,Playback(vtiger-fax)
exten => _999XX18112,5,Hangup

exten => _999XX19112,1,Wait(2)
exten => _999XX19112,2,Answer
exten => _999XX19112,3,Playback(vtiger-email)
exten => _999XX19112,4,Playback(vtiger-email)
exten => _999XX19112,5,Hangup

exten => _999XXXX112,1,Wait(5)
exten => _999XXXX112,2,Answer
exten => _999XXXX112,3,Playback(demo-instruct)
exten => _999XXXX112,4,Playback(demo-instruct)
exten => _999XXXX112,5,Hangup

exten => _999XXXXXX2,1,Wait(8)
exten => _999XXXXXX2,2,Answer
exten => _999XXXXXX2,3,Playback(demo-instruct)
exten => _999XXXXXX2,4,Hangup

exten => _999XXXXXX3,1,Set(PRI_CAUSE=1)
exten => _999XXXXXX3,2,Hangup

exten => _999XXXXXX4,1,Set(PRI_CAUSE=27)
exten => _999XXXXXX4,2,Hangup

exten => _999XXXXXX5,1,Wait(60)
exten => _999XXXXXX5,2,Hangup

exten => _999XXXXXX6,1,Wait(10)
exten => _999XXXXXX6,2,Answer
exten => _999XXXXXX6,3,Playback(demo-instruct)
exten => _999XXXXXX6,4,Hangup

exten => _999XXXXXX7,1,Wait(12)
exten => _999XXXXXX7,2,Answer
exten => _999XXXXXX7,3,Playback(demo-enterkeywords)
exten => _999XXXXXX7,4,Hangup

exten => _999XXXXXX8,1,Set(PRI_CAUSE=17)
exten => _999XXXXXX8,2,Hangup

exten => _999XXXXXX9,1,Wait(6)
exten => _999XXXXXX9,2,Answer
exten => _999XXXXXX9,3,Playback(demo-abouttotry)
exten => _999XXXXXX9,4,Hangup

exten => _999XXXXXX0,1,Wait(5)
exten => _999XXXXXX0,2,Answer
exten => _999XXXXXX0,3,Playback(vm-goodbye)
exten => _999XXXXXX0,4,Hangup

exten => 99999999999,1,Answer
exten => 99999999999,2,Playback(conf)
exten => 99999999999,3,Playback(conf)
exten => 99999999999,4,Playback(conf)
exten => 99999999999,5,Playback(conf)
exten => 99999999999,6,Playback(conf)
exten => 99999999999,7,Playback(conf)
exten => 99999999999,8,Playback(conf)
exten => 99999999999,9,Playback(conf)
exten => 99999999999,10,Playback(conf)
exten => 99999999999,11,Playback(conf)
exten => 99999999999,12,Playback(conf)
exten => 99999999999,13,Playback(conf)
exten => 99999999999,14,Hangup


[monitor]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

exten => s,1,Set(TIMEOUT(digit)=10)
exten => s,n,Set(TIMEOUT(response)=10)
exten => s,n,Set(MEETME_EXIT_CONTEXT=monitor_exit)
exten => s,n,Background(vm-extension) ; need audio prompt.
exten => s,n,WaitExten(10)

exten => i,1,Goto(monitor_exit,s,1)
exten => #,1,Goto(monitor_exit,s,1)
exten => t,1,Goto(monitor_exit,s,1)

exten => _8[0-2]XX,1,Meetme(8600${EXTEN:1},mqX) ; Listen
exten => _99[0-2]XX,1,Meetme(8600${EXTEN:2},X)  ; Barge

[monitor_exit]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

exten => _X,1,Goto(monitor,s,1)

exten => i,1,Goto(monitor,s,1)
exten => #,1,Goto(monitor,s,1)
exten => t,1,Goto(monitor,s,1)


meine sip.conf ist folgende:

[general]
context=default                 ; Default context for incoming calls
;allowguest=no                  ; Allow or reject guest calls (default is yes)
allowoverlap=no                 ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no               ; Disable all transfers (unless enabled in peers or users)
;realm=mydomain.tld             ; Realm for digest authentication
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
;domain=mydomain.tld            ; Set default domain for this host
;pedantic=yes                   ; Enable checking of tags in headers,
;tos_sip=cs3                    ; Sets TOS for SIP packets.
;tos_audio=ef                   ; Sets TOS for RTP audio packets.
;tos_video=af41                 ; Sets TOS for RTP video packets.
;maxexpiry=3600                 ; Maximum allowed time of incoming registrations
;minexpiry=60                   ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120              ; Default length of incoming/outgoing registration
;t1min=100                      ; Minimum roundtrip time for messages to monitored hosts
;notifymimetype=text/plain      ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10                    ; Default time between mailbox checks for peers
;buggymwi=no                    ; Cisco SIP firmware doesn't support the MWI RFC
;vmexten=voicemail              ; dialplan extension to reach mailbox sets the
disallow=all                    ; First disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference
allow=gsm
mohinterpret=default
mohsuggest=default
language=en                     ; Default language setting for all users/peers
relaxdtmf=yes                   ; Relax dtmf handling
trustrpid = no                  ; If Remote-Party-ID should be trusted
sendrpid = yes                  ; If Remote-Party-ID should be sent
progressinband=no               ; If we should generate in-band ringing always
;useragent=Asterisk PBX         ; Allows you to change the user agent string
;promiscredir = no              ; If yes, allows 302 or REDIR to non-local SIP address
;usereqphone = no               ; If yes, ";user=phone" is added to uri that contains
dtmfmode = rfc2833              ; Set default dtmfmode for sending DTMF. Default: rfc2833
;compactheaders = yes           ; send compact sip headers.
videosupport=no                 ; Turn on support for SIP video. You need to turn this on
;maxcallbitrate=384             ; Maximum bitrate for video calls (default 384 kb/s)
callevents=yes                  ; generate manager events when sip ua
;alwaysauthreject = yes         ; When an incoming INVITE or REGISTER is to be rejected,
;g726nonstandard = yes          ; If the peer negotiates G726-32 audio, use AAL2 packing
;matchexterniplocally = yes     ; Only substitute the externip or externhost setting if it matches
;regcontext=sipregistrations
rtptimeout=60                   ; Terminate call if 60 seconds of no RTP or RTCP activity
;rtpholdtimeout=300             ; Terminate call if 300 seconds of no RTP or RTCP activity
;rtpkeepalive=<secs>            ; Send keepalives in the RTP stream to keep NAT open
;sipdebug = yes                 ; Turn on SIP debugging by default, from
;recordhistory=yes              ; Record SIP history by default
;dumphistory=yes                ; Dump SIP history at end of SIP dialogue
;allowsubscribe=no              ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default     ; Set a specific context for SUBSCRIBE requests
notifyringing = yes             ; Notify subscriptions on RINGING state (default: no)
notifyhold = yes                ; Notify subscriptions on HOLD state (default: no)
limitonpeers = yes              ; Apply call limits on peers only. This will improve
;t38pt_udptl = yes            ; Default false
;register => 1234:[email protected]
;registertimeout=20             ; retry registration calls every 20 seconds (default)
;registerattempts=10            ; Number of registration attempts before we give up
;externip = 192.168.1.1        ; Address that we're going to put in outbound SIP
;externhost=test.test.com     ; Alternatively you can specify a domain
;externrefresh=10               ; How often to refresh externhost if
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0     ; Also RFC1918
localnet=172.16.0.0/12          ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
nat=yes                         ; Global NAT settings  (Affects all peers and users)
canreinvite=no          ; Asterisk by default tries to redirect the
;directrtpsetup=yes             ; Enable the new experimental direct RTP setup. This sets up
;rtcachefriends=yes             ; Cache realtime friends by adding them to the internal list
;rtsavesysname=yes              ; Save systemname in realtime database at registration
;rtupdate=yes                   ; Send registry updates to database using realtime? (yes|no)
;rtautoclear=yes                ; Auto-Expire friends created on the fly on the same schedule
;ignoreregexpire=yes            ; Enabling this setting has two functions:
;domain=mydomain.tld,mydomain-incoming
;domain=1.2.3.4                 ; Add IP address as local domain
;allowexternaldomains=no        ; Disable INVITE and REFER to non-local domains
;autodomain=yes                 ; Turn this on to have Asterisk add local host
;fromdomain=mydomain.tld        ; When making outbound SIP INVITEs to
jbenable = yes              ; Enables the use of a jitterbuffer on the receiving side of a
jbforce = no                ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize = 100             ; Max length of the jitterbuffer in milliseconds.
jbresyncthreshold = 1000    ; Jump in the frame timestamps over which the jitterbuffer is
jbimpl = fixed              ; Jitterbuffer implementation, used on the receiving side of a SIP
jblog = no                  ; Enables jitterbuffer frame logging. Defaults to "no".
qualify=yes             ; By default, qualify all peers at 2000ms
limitonpeer = yes       ; enable call limit on a per peer basis, different from limitonpeers

#include sip-vicidial.conf

; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:[email protected]:5060
;
; setup account for SIP trunking:
; [SIPtrunk]
; disallow=all
; allow=ulaw
; allow=alaw
; type=friend
; username=testSIPtrunk
; secret=test
; host=10.10.10.16
; dtmfmode=inband
; qualify=1000
als dialplan bezeichnet vicidial den bereich indem ich die extensions eingebe so wie in meiner ersten mail gepostet.

zitat:
Außerdem ist Dein Dial falsch, ausgeschrieben würde das bei Dir Dial(SIP/viva/${EXTEN:1}) ergeben.
Die Syntax für abgehende Anrufe über SIP ist Dial(SIP/${EXTEN:1}@viva)
Das lässt sich mit der globalen Variable so nicht darstellen.

soll das heissen mein global string ist falsch ? oder meine exten?

soll ich das kleine t und das grosse T rausnehmen ?
Die Option t solltest Du bei abgehenden Gesprächen raus nehmen.
 
aber die exten ist passend den ich waehle ein 9 davor und dann 10 ziffern so wie in der exten:

exten => _9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXX,2,Dial(${VIVA}/${EXTEN:1},,tTo)
exten => _9XXXXXXXXXX,3,Hangup

wenn das deiner meinung nach falsch ist... wie muss den die richtige aussehen ??
 
Mag schon sein, dass die exten passt, aber sie befindet sich nicht in dem Context, den Dein Tool zum Wählen verwenden will.

Ist das was Du als "dialplan" bezeichnest der Inhalt von [default] oder ein anderer Context?

Also wie heißt denn bitte der Context, in dem das drin steht? Wahrscheinlich nicht [default].

Zur Veranschaulichung:
Code:
[default]
bei Dir entweder leer oder nicht vorhanden oder irgendwas drin

[keineahnungwiederheißt]
exten=>_9XXXXXXXXXX,1,blablub
...

Wenn Dein Asterisk nicht über SIP von extern erreichbar ist (und auch nur dann!) könntest Du das mit
Code:
[default]
include => keineahnungwiederheißt
beheben.

Dann ist aber immer noch Dein Dial falsch, siehe oben.
 
Zuletzt bearbeitet von einem Moderator:
in der extensions.conf steht als default:
Code:
[default]
include => vicidial-auto

; VICI-GROUP DIRECT SUPPORT LINE (VICIHELP[84244357])
exten => _84244XXX,1,Dial(IAX2/vicihelp/${EXTEN:5})

; Local agent alert extensions
exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
; Local blind monitoring
exten => _08600XXX,1,Dial(${TRUNKblind}/6${EXTEN:1},55,To)


; playback of recorded prompts
exten => _851XXXXX,1,Answer
exten => _851XXXXX,2,Playback(${EXTEN})
exten => _851XXXXX,3,Hangup

; this is used for playing a message to an answering machine forwarded from AMD in VICIDIAL
exten => _7851XXXXX,1,WaitForSilence(2000,2) ; AMD got machine.  leave message after recording
exten => _7851XXXXX,2,Playback(${EXTEN:1})
exten => _7851XXXXX,3,AGI(VD_amd_post.agi,${EXTEN:1})
exten => _7851XXXXX,4,Hangup


meine extensions.conf verweist aber auch auf die vicidial.extensions.conf 
#include extensions-vicidial.conf

dort steht das was ich in vicidial eintrage:

[vicidial-auto]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

; Local Server: 192.168.1.2
exten => _192*168*001*002*.,1,Goto(default,${EXTEN:16},1)
; VICIDIAL Carrier: SIPEXAMPLE -  SIPEXAMPLE 
exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:2}@viva)
exten => _91XXXXXXXXXX,3,Hangup
 
[Sep 21 16:43:24] == Starting Local/8******@default-d14b,1 at default,91*********,1 failed so falling back to exten 's'
sagt eindeutig, er findet kein passendes Pattern.

Erst schreibst Du
exten=>_9XXXXXXXXXX
was eigentlich zu der gewählten Nummer passen würde.

Jetzt hast Du plötzlich
exten=>_91XXXXXXXXXX
was wiederum eine Stelle zu viel ist für die Nummer.

Ändere das in
exten=>_9XXXXXXX.
dann passt es auf jeden Fall.
 
die 9 und die 91 werden sowieso nicht mituebermittelt da ich EXTEN:1 oder EXTEN:2 dann auch aendere. war nur ein versuch.

ich habe jetzt das ganze in exten=>_9XXXXXXX. geaendert also so:

exten => _9XXXXXXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXX.,2,Dial(SIP/${EXTEN:1}@viva)
exten => _9XXXXXXX.,3,Hangup


und erhalte immer noch folgendes cli:
Code:
[Sep 23 04:47:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 23 04:47:07]     -- Executing [96959791530@default:1] AGI("SIP/cc103-0000004d", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 23 04:47:07]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 23 04:47:07]     -- Executing [96959791530@default:2] Dial("SIP/cc103-0000004d", "SIP/6959791530@viva") in new stack
[Sep 23 04:47:07]     -- Called 6959791530@viva
[Sep 23 04:47:07]     -- Got SIP response 486 "Busy Here" back from 83.235.24.86
[Sep 23 04:47:07]     -- SIP/viva-0000004e is busy
[Sep 23 04:47:07]   == Everyone is busy/congested at this time (1:1/0/0)
[Sep 23 04:47:07]     -- Executing [96959791530@default:3] Hangup("SIP/cc103-0000004d", "") in new stack
[Sep 23 04:47:07]   == Spawn extension (default, 96959791530, 3) exited non-zero on 'SIP/cc103-0000004d'
[Sep 23 04:47:07]     -- Executing [h@default:1] DeadAGI("SIP/cc103-0000004d", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----17-----BUSY----------") in new stack
[Sep 23 04:47:07]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----17-----BUSY---------- completed, returning 0
go*CLI>


mit der konfiguration:
Code:
[viva]
type=peer
username=302117706001
secret=695809
fromuser=302117706001
host=dynamic
dtmfmode=rfc2833
fromdomain=voip.viva.gr
context=trunkinbound
nat=no
dtmfmode=rfc2833
disallow=all
allow=g729
allow=ulaw
allow=alaw
canreinvite=no
allow=all
qualify=1000 

global string: VIVA=SIP/viva

dialplan:
exten => _9XXX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXX.,2,Dial(${SIPTRUNK}/${EXTEN:1},,tTor)
exten => _9XXX.,3,Hangup 


hoere ich das es wahlt aber das gespraech kommt nie an. 

mein cli waehrend des anrufs...

[Sep 23 05:11:08]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 23 05:11:09]     -- Executing [96959791530@default:1] AGI("SIP/cc103-0000005c", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 23 05:11:09]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 23 05:11:09]     -- Executing [96959791530@default:2] Dial("SIP/cc103-0000005c", "SIP/tata-out/6959791530||tTor") in new stack
[Sep 23 05:11:09]     -- Called tata-out/6959791530
go*CLI>

die nummer die gewaehlt werden soll ist die 6959791530
9 ist dialprefix welches mit EXTEN:1 nicht mitgewaehlt wird.

bin am verzweifeln....
 
die 9 und die 91 werden sowieso nicht mituebermittelt da ich EXTEN:1 oder EXTEN:2 dann auch aendere.

Deshalb muss aber doch vorher das Pattern passen! Was Du nachher dem Dial übergibst ist erst mal unerheblich, wenn Du vorher schon gar kein match im Dialplan bekommst.

Aber gut, zumindest wird jetzt mal gewählt. Jetzt passt dem Provider die gewählte Nummer nicht oder Du hast ein Konfigurationsproblem des SIP peers. Wobei Deine sip.conf eigentlich soweit gut aussieht. :noidea:
 
ich habe auf nat=no gewechselt und host=voip.viva.gr

jetzt ist der peer sichtbar...

cc103/cc103 192.168.1.59 D N 8778 OK (112 ms)
cc102/cc102 (Unspecified) D N 0 UNKNOWN
cc101/cc101 (Unspecified) D N 0 UNKNOWN
cc100/cc100 (Unspecified) D N 0 UNKNOWN
viva/302117706001 83.235.24.86 5060 OK (22 ms)

allerdings hoere ich immer noch nur ein waehlen...
 
Ich möchte mal behaupten
exten => _9XXX.,2,Dial(${SIPTRUNK}/${EXTEN:1},,tTor)
muss deswegen aber immer noch
exten => _9XXX.,2,Dial(SIP/${EXTEN:1}@viva,,T)
heißen.

Das Freizeichen kommt durch die Option r im Dial, nicht von der Gegenstelle.
 
mit dieser einstellung bekomme ich dann wieder diese meldung
Code:
[Sep 23 06:19:07]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 23 06:19:08]     -- Executing [96959791530@default:1] AGI("SIP/cc103-00000070", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 23 06:19:08]     -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 23 06:19:08]     -- Executing [96959791530@default:2] Dial("SIP/cc103-00000070", "SIP/6959791530@viva||T") in new stack
[Sep 23 06:19:08]     -- Called 6959791530@viva
[Sep 23 06:19:08]     -- Got SIP response 486 "Busy Here" back from 83.235.24.86
[Sep 23 06:19:08]     -- SIP/viva-00000071 is busy
[Sep 23 06:19:08]   == Everyone is busy/congested at this time (1:1/0/0)
[Sep 23 06:19:08]     -- Executing [96959791530@default:3] Hangup("SIP/cc103-00000070", "") in new stack
[Sep 23 06:19:08]   == Spawn extension (default, 96959791530, 3) exited non-zero on 'SIP/cc103-00000070'
[Sep 23 06:19:08]     -- Executing [h@default:1] DeadAGI("SIP/cc103-00000070", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----17-----BUSY----------") in new stack
[Sep 23 06:19:08]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----17-----BUSY---------- completed, returning 0
go*CLI>
 

Zurzeit aktive Besucher

Keine Mitglieder online.

Statistik des Forums

Themen
246,300
Beiträge
2,249,713
Mitglieder
373,904
Neuestes Mitglied
Elemir
Holen Sie sich 3CX - völlig kostenlos!
Verbinden Sie Ihr Team und Ihre Kunden Telefonie Livechat Videokonferenzen

Gehostet oder selbst-verwaltet. Für bis zu 10 Nutzer dauerhaft kostenlos. Keine Kreditkartendetails erforderlich. Ohne Risiko testen.

3CX
Für diese E-Mail-Adresse besteht bereits ein 3CX-Konto. Sie werden zum Kundenportal weitergeleitet, wo Sie sich anmelden oder Ihr Passwort zurücksetzen können, falls Sie dieses vergessen haben.